Electronic filters, hearing aids and methods

ABSTRACT

An electronic filter for an electroacoustic system. The system has a microphone for generating an electrical output from external sounds and an electrically driven transducer for emitting sound. Some of the sound emitted by the transducer returns to the microphone means to add a feedback contribution to its electrical output. The electronic filter includes a first circuit for electronic processing of the electrical output of the microphone to produce a first signal. An adaptive filter, interconnected with the first circuit, performs electronic processing of the first signal to produce an adaptive output to the first circuit to substantially offset the feedback contribution in the electrical output of the microphone, and the adaptive filter includes means for adapting only in response to polarities of signals supplied to and from the first circuit. Other electronic filters for hearing aids, public address systems and other electroacoustic systems, as well as such systems and methods of operating them are also disclosed.

GOVERNMENT SUPPORT

This invention was made with U.S. Government support under VeteransAdministration Contracts V674-P-857 and V674-P-1736 and NationalAeronautics and Space Administration (NASA) Research Grant No.NAG10-0040. The U.S. Government has certain rights in this invention.

CROSS-REFERENCE TO RELATED APPLICATIONS

The present application is a continuation of application Ser. No.07/792,706, for "Electronic Filters, Repeated Charge RedistributionSignal Conversion Apparatus, Hearing Aids and Methods" filed Nov. 15,1991 by Robert E. Morley, Jr., A. Maynard Engebretson (the inventorherein), George L. Engel and Thomas J. Sullivan, now U.S. Pat. No.5,225,836, which is a divisional of application Ser. No. 07/180,170filed Apr. 11, 1988 now issued as U.S. Pat. No. 5,111,419 which is acontinuation-in-part of application Ser. No. 07/172,266 filed Mar. 23,1988 now issued as U.S. Pat. No. 5,016,280, all of which applicationsare incorporated by reference herein.

NOTICE

Copyright ©1988 Central Institute for the Deaf. A portion of thedisclosure of this patent document contains material which is subject tocopyright protection. The copyright owner has no objection to thefacsimile reproduction by anyone of the patent document or the patentdisclosure, as it appears in the Patent and Trademark Office patent fileor records, but otherwise reserves all copyright rights whatsoever.

FIELD OF THE INVENTION

The present invention relates to electronic filters for hearing aids,public address systems, and other electroacoustic systems, hearing aidswith electronic filters and methods of operation. More particularly, thepresent invention relates to electronic adaptive filters and filteringmethods to offset disadvantageous acoustic feedback in hearing aids,public address systems and other electroacoustic systems.

BACKGROUND OF THE INVENTION

Electronic hearing aids and methods are discussed in coassigned U.S.Pat. No. 4,548,082 which is incorporated herein by reference as anexample of an electroacoustic system in which the present invention canbe used.

Without limiting the scope of the present invention in hearing aids,public address systems and electroacoustic systems in general, thebackground of the invention is described specifically in connection withits application to hearing aids.

A person's ability to hear speech and other sounds well enough tounderstand them is clearly important in employment and many other dailylife activities. Improvements in hearing aids which are intended tocompensate or ameliorate hearing deficiencies of hearing impairedpersons are consequently important not only to these persons but also tothe community at large.

Unfortunately, presently available hearing aids are often subject to afeedback phenomenon which produces distortion, ringing, and squealing.If the hearing aid can be adjusted by the hearing impaired person toreduce the feedback, the volume delivered is usually reduced also. Notonly is the person enmeshed in a dilemma of choosing between feedbackand reduced volume, but also the adjustment process itself is one moreinconvenience in that person's life. Not surprisingly, many users ofhearing aids leave the aid misadjusted or simply put the hearing aidaside because of the inconvenience and lack of satisfactory operationwith which they must cope.

Conventionally, a microphone in the hearing aid generates an electricaloutput from external sounds. An amplifying circuit in the aid provides afiltered version of the electrical output corresponding to the soundspicked up by the microphone. The filtering can be due to an inherentcharacteristic of the amplifying circuit or may be deliberatelyintroduced. The amplified and filtered output of the hearing aid is fedto an electrically driven "receiver" for emitting sound into the ear ofthe user of the hearing aid. (In the hearing aid field, a receiver isthe name of an electronic element analogous to a loudspeaker or otherelectroacoustic transducer.) Some of the sound emitted by the receiverreturns to the microphone to add a feedback contribution to theelectrical output of the microphone. The feedback is amplified by thehearing aid, and ringing or squealing often arise in an endlesslycircular feedback process.

SUMMARY OF THE INVENTION

Among the objects of the present invention are to provide improvedelectronic filters, hearing aids, other electroacoustic systems andmethods which substantially prevent undesirable feedback ringing andsquealing; to provide improved electronic filters, hearing aids andother electroacoustic systems which are reliable, compact andeconomical; to provide improved electronic filters, hearing aids andmethods which reliably reject undesirable feedback which otherwise couldoccur in the daily life of a hearing aid user as when the user puts ahand near the ear or the ear is near a chair back or wall; and toprovide improved electronic filters, hearing aids and methods whichreliably reject undesirable feedback when the user's jaw moves intalking or chewing.

Generally, one form of the invention is an electronic filter for ahearing aid. The hearing aid has a microphone for generating anelectrical output from sounds external to a user of the hearing aid andan electrically driven receiver for emitting sound into the ear of theuser. Some of the sound emitted by the receiver returns to themicrophone to add a feedback contribution to its electrical output. Theelectronic filter includes an electronic processor for processing theelectrical output of the microphone to produce a first signal and forcombining the first signal with a second distinct signal for thereceiver. An adaptive filter, interconnected with the electronicprocessor processes the first signal and second distinct signal whilethe electronic processor is producing the first signal to produce anadaptive output to the electronic processor to continuouslysimultaneously offset the feedback contribution in the electrical outputof the microphone.

Generally and in another form of the invention, a hearing aid includes amicrophone for generating an electrical output from sounds external to auser of the hearing aid and an electrically driven receiver for emittingsound into the ear of the user of the hearing aid. Some of the soundemitted by the receiver returns to the microphone to add a feedbackcontribution to its electrical output. An electronic processor processesthe electrical output of the microphone to produce a first signal andcombines the first signal with a second distinct signal for the receiverof the hearing aid. A controller varies the second distinct signal inmagnitude as a function of the magnitude of the first signal. Anadaptive filter is interconnected with the electronic processor forprocessing the first signal and second distinct signal to produce anadaptive output to the electronic processor to substantially offset thefeedback contribution in the electrical output of the microphone in thehearing aid.

In general, and in a further form of the invention, a hearing aidincludes a microphone for generating an electrical output from soundsexternal to a user of the hearing aid and an electrically drivenreceiver for emitting sound into the ear of the user of the hearing aid.Some of the sound emitted by the receiver returns to the microphone toadd a feedback contribution to its electrical output, which feedback isto be substantially offset. An electronic processor processes theelectrical output of the microphone to produce a first signal. A firstsignal combiner combines the first signal with a second distinct signalfor the receiver of the hearing aid in a proportion wherein themagnitude of the second distinct signal is generally less than themagnitude of the first signal at loudness of normal speach. A secondsignal combiner combines the first signal with the second distinctsignal to produce a control signal for the electronic processor whereinthe magnitude of the second distinct signal is generally greater thanthe magnitude of the first signal at loudness of normal speach.

Generally, still another form of the invention includes an electronicfilter for a hearing aid. The hearing aid has a microphone forgenerating an electrical output from sounds external to a user of thehearing aid and an electrically driven receiver for emitting sound intothe ear of the user. Some of the sound emitted by the receiver returnsto the microphone to add a feedback contribution to its electricaloutput, which feedback is to be substantially offset. The electronicfilter includes a signal generator for generating a probe signal for thereceiver of the hearing aid so that a sound corresponding to the probesignal is included in the sound emitted by the receiver. An electronicprocessor processes the probe signal in accordance with a series ofcoefficients to produce a first output. A signal combiner combines thefirst output with the electrical output from the microphone to produce acombined signal input having a changing polarity. Some of the soundcorresponding to the probe signal returns to the microphone. A signalprocessor electronically derives a series of values having polaritiesresponsive to the probe signal. A first register holds running totalsand a second register holds the series of coefficients. A circuitincreases and decreases each running total in the first registerdepending on whether a corresponding value in the series of values hasthe same or opposite polarity compared to the combined signal input. Anadder respectively adds the running totals in the first register to thecoefficients in the second register to update the coefficients for theelectronic processor less frequently than said circuit increases anddecreases the running totals in the first register.

In general, yet another form of the invention is a hearing aid adaptedto be coupled to a user. The hearing aid has a microphone for generatingan electrical output from sounds external to the user and anelectrically driven receiver for emitting sound into the ear of theuser. Some of the sound emitted by the receiver returns to themicrophone to add a feedback contribution to its electrical output. Anelectronic processor processes the electrical output of the microphoneto produce a first signal for the receiver. An adaptive filter has aninput coupled to the first signal. The adaptive filter processes thefirst signal while the electronic processor is producing the firstsignal to continuously simultaneously offset the feedback contributionin the electrical output of the microphone.

In general, a yet further form of the invention is an electronic filterfor an electroacoustic system. The system has a microphone forgenerating an electrical output from external sounds and an electricallydriven transducer for emitting sound. Some of the sound emitted by thetransducer returns to the microphone to add a feedback contribution toits electrical output. The electronic filter includes a VLSI die and anelectronic processor fabricated on the VLSI die for processing theelectrical output of the microphone to produce a first signal. Anall-hardware adaptive filter united with and connected to the electronicprocessor on the VLSI die processes the first signal while theelectronic processor is producing the first signal to produce anadaptive output to the electronic processor to substantially offset thefeedback contribution in the electrical output of the microphone.

Generally, a still further form of the invention is a method ofoperating a hearing aid having a microphone for generating an electricaloutput from sounds external to a user of the hearing aid and anelectrically driven receiver for emitting sound into the ear of the userof the hearing aid. Some of the sound emitted by the receiver returns tothe microphone to add a feedback contribution to its electrical output.The method includes the steps of processing the electrical output of themicrophone to produce a first signal and combining the first signal witha second distinct signal for the receiver of the hearing aid. The methodfurther includes the steps of adaptively filtering the first signal andsecond distinct signal during the processing step to produce an adaptiveoutput and feeding back the adaptive output to continuouslysimultaneously offset the feedback contribution in the electrical outputof the microphone for the processing step.

Other forms of the invention are also disclosed, including hearing aidand other system combinations and methods for operating them and otherelectroacoustic systems.

Other objects and features will be in part apparent and in part pointedout hereinafter.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 is a pictorial of a user with a hearing aid of the inventionincluding an electronic filter according to the invention, part of thehearing aid shown in cross-section;

FIG. 2 is a pictorial side view of the hearing aid of FIG. 1;

FIG. 3 is an electrical block diagram of the hearing aid of FIG. 1including an inventive electronic filter circuit with an inventiveall-hardware circuit, also showing an acoustic feedback path;

FIG. 4 is an electrical block diagram of an inventive all-hardwarecircuit for the electronic filter of FIG. 3;

FIG. 5 is an electrical block diagram of a serial interface, randomaccess memory and several filter-limit-filter sections in the electronicfilter of FIG. 3;

FIG. 6 is a partially graphical, partially block diagrammaticrepresentation of a finite impulse response (FIR) digital filter used inthe some of the inventive circuits;

FIG. 7 is partially graphical, partially block diagrammatic illustrationof operations in a preferred embodiment for making a digital filteradaptively simulate or mimic the characteristics of a feedback path;

FIG. 8 is a diagram of a feedback path supplied with a probe signalwherein the feedback path has a dispersion that offers many differentdelays to the same probe signal;

FIG. 9 is three graphs of voltage versus time, the first two graphsshowing a combined input signal and a probe signal in synchronism, andthe third graph showing a product signal being persistently positive forincreasing a filter coefficient for simulating the feedback path;

FIG. 10 is another set of three graphs of voltage versus time, the firsttwo graphs showing the combined input signal delayed relative to theprobe signal, and the third graph showing a product signal sometimespositive and sometimes negative and thus having negligible effect on thevalue of another filter coefficient for simulating the feedback path;

FIG. 11 is a pair of graphs of voltage versus time showing that thepolarity of a combined input signal including speech corresponds inpolarity to a pseudorandom noise probe signal near zero crossings of thespeech;

FIG. 12 is a partially block, partially schematic diagram of aninventive logic circuit for controlling an adaptive filter in FIG. 4;

FIG. 13 is a schematic diagram of a data bus carrying a digital signalhaving a magnitude and a polarity;

FIG. 14 is a schematic diagram of an analog signal line with acomparator for detecting a polarity signal;

FIG. 15 is another block diagram of an all hardware circuit on a VLSIdie having first and second combining circuits for the noise signalaccording to another inventive embodiment;

FIG. 16 is a partially block, partially schematic diagram of anotherinventive form of logic circuit for controlling an adaptive filter forsimulating a feedback path;

FIG. 17 is a block diagram of another inventive all-hardware circuit ona VLSI die;

FIG. 18 is a graph of speech and pseudorandom probe noise componentsversus time in a combining circuit wherein the magnitude of the noise,as a second distinct signal, is generally greater than the magnitude ofthe filtered signal at loudness of normal speech;

FIGS. 19 and 20 are a pair of graphs of speech and pseudorandom probenoise components versus time in another combining circuit wherein themagnitude of the noise, as a second distinct signal, is generally lessthe magnitude of the filtered signal at loudness of normal speech, andthe magnitude of the noise is varied in magnitude as a function of themagnitude of the speech;

FIG. 21 is a block diagram of an inventive all-hardware circuit on aVLSI die where a subtracter is used as a combining circuit foroffsetting the feedback contribution;

FIG. 22 is a partially block, partially schematic diagram of aninventive logic circuit for use in the circuit of FIG. 21;

FIG. 23 is a partially block, partially schematic diagram of aninventive logic circuit for producing running totals and adding them tocoefficients for a digital filter from time to time;

FIG. 24 is a partially block, partially schematic diagram of anotherinventive logic circuit for producing running totals and adding them tocoefficients for a digital filter from time to time;

FIG. 25 is a graph of decibels (dB) versus frequency (kiloHertz, kHz.)showing a frequency spectrum (in solid) of the output of a hearing aidwithout feedback offset compared to a frequency spectrum (in dashed) ofthe output of a hearing aid with feedback offset according to theinvention;

FIG. 26 is a block diagram of another inventive electronic filter for ahearing aid according to the invention improved with an infinite impulseresponse adaptive filter;

FIG. 27 is a block diagram of yet another inventive electronic filterfor a hearing aid according to the invention improved with an infiniteimpulse response adaptive filter and an error filter; and

FIG. 28 is a process flow diagram illustrating some inventive methodsfor operating an electronic filter and a hearing aid of the invention.

Corresponding reference characters indicate corresponding partsthroughout the several views of the drawings.

DETAILED DESCRIPTION OF PREFERRED EMBODIMENTS

In FIG. 1 a hearing aid 11 receives external sounds at an inputmicrophone 13 in an earpiece 14. Microphone 13 generates an electricaloutput from sounds external to the user of the hearing aid for anover-the-ear-unit 15 which produces an electrical output for a receiveror transducer 17 that emits filtered and amplified sound from earpiece14 into the ear of the user of the hearing aid. (In another hearing aid,not shown, the microphone 13 and receiver 17 are in a behind-the-ear(BTE) unit and not in an earpiece, and the improvements described hereinare equally applicable to this and other units.)

For purposes of the present disclosure it is important to note that someof the sound emitted by receiver 17 returns to the microphone 13 asfeedback indicated by arrows such as 19 and 21 from the opening of achannel 23 by which receiver 17 communicates with the ear canal of theuser. Other feedback passes through the side of earpiece 14 and takes ashorter path to the input microphone as illustrated by arrow 25. Somesound 29 feeds back directly from receiver 17 through interior absorbermaterial 27 of earpiece 14 to the microphone 13.

Feedback is disadvantageously associated with squealing, ringing,erratic operation and instability in the operation of hearing aid 11.Accordingly it is desirable to find some way to permit hearing aid 11 tooperate more satisfactorily even though feedback according to thenumerous paths indicated by arrows 19, 21, 25 and 29 unavoidably occurs.

FIG. 2 shows a side view of hearing aid 11 with its over-the-ear unit 15which includes filtering and amplifying circuitry. In a clinical fittingprocedure, unit 15 is loaded with digital information through aconnector 35 connected by a cord 37 from a host computer system such asthat described in U.S. Pat. No. 4,548,082 which is incorporated hereinby reference. After the hearing aid 11 has been loaded at the clinicwith information which suits it to ameliorate the particular hearingdeficiency of the user, connector 35 is detached from the rest of thehearing aid and replaced with a battery pack 39 for convenient dailyuse. One type of preferred embodiment is improved with inventivefeedback offsetting circuitry that requires no additional informationfrom the host system to govern the offsetting operations.

In FIG. 3 a block diagram of unit 15 includes an electronic filter 41that has an all-hardware circuit 43 that accomplishes numerous functionsfor ameliorating the hearing deficiency of the user and offsets feedbackcontribution 44 which unavoidably occurs in hearing aid 11 as alreadydiscussed. Circuit 43 is discussed in greater detail hereinbelow.

Also as shown in FIG. 3, connector 35 of FIG. 2 includes terminals for apair of serial lines from the host computer system, a line for externalsupply voltage V and a DC return common line. A battery in connector 39of FIG. 2 is shown in FIG. 3. The serial lines of connector 35 from thehost system as an external source are connected to a serial interfaceintegrated circuit 45 in unit 15. Serial interface circuit 45 providesthe received filter coefficients in parallel form along a data bus 47 toall-hardware circuit 43. A random access memory (RAM) 49 holdscoefficient data which have been downloaded from the host system. RAM 49is supplied with voltage V through a diode 51. When the externalconnector 35 is being replaced with connector 39, RAM 49 retains thecoefficients by using temporary power from a small battery 53.

All-hardware circuit 43 processes digitized audio information from inputmicrophone 13. Microphone 13 is suitably an electret microphone whichreceives battery voltage V for operating power and provides an output toan anti-aliasing low pass filter (LPF) 59 which has a cutoff frequencyof 6 kilohertz. LPF 59 in turn feeds a sample and hold circuit S/H 61.S/H 61 in turn feeds analog samples to an analog-to-digital converter(ADC) 63 which converts the samples to digital form and temporarilyholds them in an ADC output register. The samples are provided from theADC register to the all-hardware circuit 43. Circuit 43 processes themand supplies a digital output, corresponding to filtered and amplifiedexternal sound with feedback offset, to a digital to analog converter(DAC) 65. The analog output of DAC 65 is fed to an analog active filter(AAF) 67 which amplifies the analog output and drives receiver 17through a blocking capacitor 69. Receiver 17 produces soundcorresponding to the analog output for the ear as described inconnection with FIG. 1. Unavoidable feedback to the input microphone 13is indicated by curved feedback arrow 44 of FIG. 3.

All-hardware circuit 43 of FIG. 3 has an associated clock and decodercircuit 81 and a control latch 83. Circuit 81 provides clock pulses toall-hardware circuit 43 on a line 85 and various control pulses foroperating serial interface 45 via a line 87. Sample and hold circuit 61is controlled via a line 89 from control latch 83, which in turn iscontrolled by circuit 81 through a decoder output line 91. ADC 63 iscontrolled via a line 93 from control latch 83. Circuit 81 also clocksthe output register REG of ADC 63 via a line 95. Circuit 81 furtherclocks DAC 65 through a line 97. The details of the control circuitry,as well as of a power-on reset circuit POR, are suitably provided in aconventional manner and need not be discussed further herein.

In FIG. 4 further details of the all-hardware circuit 43 areillustrated, together with blocks indicating the transfer functions ofthe microphone 13 and receiver 17.

External sound Xi is converted by microphone 13 and circuits 61 and 63of FIG. 3 to digital electrical form for circuit 43 according to atransfer function Hm indicated by block 101 of FIG. 4. A furthertransfer function Hr indicated by block 103 represents conversion by DAC65, additional amplification by AAF 67 beyond that provided by circuit43, and transducer action by receiver 17. Part of a resulting acousticoutput Xo from receiver 17 is fed back according to another transferfunction Hf block 105 and inherently acoustically summed at the inputmicrophone 13 (Hm 101). Accordingly feedback contribution 44 is mingledwith the external sound Xi when it reaches all-hardware circuit 43.

The digital output of block 101 is fed to a combining circuit such as asummer 107 of circuit 43. The output of combining circuit 107 isdesignated En and is supplied to a digital filter 109 operatingaccording to the data in RAM 49 of FIG. 3 to ameliorate the hearingdeficiencies of the user. This digital filter 109 has a transferfunction Hs. The output of filter 109 constitutes a filtered signalwhich is supplied to an additional combining circuit 111 that producesan output Y to drive the Hr block 103. The circuitry with elements 107,109 and 111 is an example of a first means (107, 109, 111) forelectronic processing of the electrical output of the microphone toproduce a filtered signal and for combining the filtered signal with asecond distinct signal for the receiver of the hearing aid. Filter 109and combining circuit 111 also act as an example of a means foramplifying the electrical output of the microphone to produce anamplified signal and for combining the amplified signal with apseudorandom noise signal to produce a combined amplified signal for thereceiver.

In this preferred embodiment of FIG. 4 the output Y is further also fedback through a special adaptive filter 113, with transfer function He,which continually and variably further filters the output Y and producesa resulting output of its own for the combining circuit 107. The outputof adaptive filter 113 thereby substantially offsets the feedbackcontribution 44 from block 105 even though not only the level but alsothe transfer function Hf of that feedback is changing with time.

For example, when the user sits near a wall the feedback transferfunction Hf is different from Hf when the user is in the center of alarge room. The user also changes the feedback path and thus Hf bymoving a hand near the ear, or by chewing or otherwise moving the jaw orin numerous other ways, in the daily life routine. Remarkably theadaptive filter 113 offsets this varying feedback in a manner whichconsiderably increases the convenience and value of the hearing aid tothe user.

Control for adaptive filter 113 involves a pseudorandom noise generator115 and a logic circuit 117. Noise generator 115 produces a digitalpseudorandom noise signal Se, which is supplied to combining circuit 111and there weighted by an appropriate weight W1 (or multiplying factor)for the pseudorandom noise signal so that the output of combiningcircuit 111 is

    Y=Hs*En+W1·Se                                     (1)

where Y is the output of circuit 111, En is the combined signal inputprocessed by the filter function Hs, W1 is a weight multiplied bypseudorandom noise Se, which is a function of sample number n. Asterisk"*" refers to convolution, as discussed hereinbelow in connection withFIG. 6.

The Y output of combining circuit 111 is provided to logic circuit 117which has as an additional input a single one of the bits of inputsignal En. Logic circuit 117 acts as a controller for the adaptivefilter 113, which remarkably varies the transfer function He of filter113 to equal (or mimic) the feedback loop transfer function HrHfHm as Hfvaries. In this way the feedback can be offset at all times by theoutput of filter 113 to combining circuit 107.

Advantageously, pseudorandom noise Se is thus supplied to the hearingaid circuitry at combining circuit 111 as a second electrical signal (orprobe signal) that probes the feedback path Hf 105 since some soundcorresponding to the probe signal is emitted by receiver 17 along withamplified speech. Also, the noise is acoustically fed back to themicrophone 13 with the rest of feedback contribution 44 and becomesincluded in signal En to which the logic circuitry 117 that controlsfilter 113 responds.

The block diagram of FIG. 4 is illustrative of both inventive apparatusand methods for use in hearing aid applications and is also applicablein systems generally in which instability occurs because of undesirablecoupling between output and input transducers, such as often occurs inpublic address systems.

Key parts of the apparatus and process are illustrated in FIG. 4 andinclude excitation source Se 115 and adaptive equalization filter 113with transfer function He. Other parts in FIG. 4 represent the inputsignal Xi, input transducer and circuit Hm 101, system process Hs 109,output circuit and transducer Hr 103, output signal Xo, and undesirableexternal feedback path with transfer function Hf 105. The adaptiveprocess is such that the error signal En is minimized in the least meansquare sense by adjusting the coefficients of the filter transferfunction He to offset or cancel the presence of the external feedbackpath, assuming a zero mean signal at the output of Hm. Adjustment of thefilter coefficients in filter 113 of FIG. 4 is accomplished according tothe recursive expression:

    C.sub.i (n+l)=C.sub.i (n)+λ·sgn[En·Y(n-i)](2)

where C_(i) (n+l) represents the ith coefficient at sample number n+l,C_(i) (n) represents the ith coefficient at sample number n, Enrepresents the error (also called combined signal input herein) atsample number n, and Y(n-i) represents the input to the filter at samplenumber n-i. The second term on the right of the equation adds orsubtracts a positive constant λ (lambda) to each ith coefficientdepending on the sign of the product term. In this way the coefficientsare adjusted over time to minimize the mean square of the error En,which is primarily the difference between the signals due to theexternal feedback path and the internal adaptive filtering path (alsocalled an equalization path herein) if the external input Xi be ignored.The influence of the externally derived signal Xi is minimized by thecombined signal delay through system process Hs and either feedback pathHf or adaptive filter path He. The system readily accommodates theaddition of extra delay in system process Hs or in series with systemprocess Hs as desired to even further reduce any effect of the externalsignal Xi.

The recursive expression in the above equation (2) is uncomplicated andis economically implemented in VLSI (very large scale integration) form.Calculation of the sign of the product term is readily accomplished withan exclusive-or logical operation on the polarity information in errorsignal En and each sample of output Y represented by Y(n-i) going backin time from latest value n (i=0). When lambda is made to be unity,updating each coefficient is an up-down counter operation that iscontrolled by the sign of the product term. The rate of adaption isdetermined by the recursion rate, or clock rate in the electronicembodiments.

In FIG. 4, circuit elements 107, 109, 111, 113, 115 and 117 areimplemented in VLSI on a VLSI die or substrate 121. A discussion of VLSIprinciples which is generally available to workers in the field is foundin C. A. Meade, Introduction to VLSI Systems, Addison-Wesley, 1980, seefor instance pages 60-84, 91-115, 155-164 and Plates 1-15. Systemprocess Hs 109 is implemented as a first all-hardware digital filter incombination with summer 107 and combining circuit 111 for electronicprocessing of the electrical output of the microphone in accordance witha first series of digital coefficients to produce a filtered signal andfor combining the filtered signal with a second distinct signal for thereceiver. Adaptive filter He 113 is also implemented as a secondall-hardware digital filter united with and connected to the firstall-hardware digital filter as one VLSI circuit on die or substrate 121,for electronic processing of the filtered signal and second distinctsignal in accordance with a second series of digital coefficients toproduce an adaptive output. The adaptive output is combined with themicrophone output, to produce a combined signal input for the firstdigital filter, by combining circuit 107 which is suitably also laiddown on die 121 with such other elements in FIG. 3 as maximizeconvenience, economy and reliability. Further, circuit 117 isimplemented as an all-hardware logic circuit united with and connectedto the first all-hardware digital filter 109 for adaptively varying thedigital coefficients of the second all-hardware digital filter 113 as afunction of the combined signal input so that the adaptive outputsubstantially offsets the feedback contribution in the electrical outputof the microphone.

Adaptive filter 113 is thus an example of a digital adaptive filtermeans, interconnected with the first means, for electronic processing ofthe filtered signal and second distinct signal to produce an adaptiveoutput to the first means (107, 109, 111) to substantially offset thefeedback contribution in the electrical output of the microphone meansin the hearing aid, the digital adaptive filter means including meansfor adapting only in response to polarities of signals supplied to andfrom said first means.

In logic circuit 117, the coefficient updating for filter 113 issuitably accomplished by implementing an adder-subtractor in which casethe positive integer lambda λ is added to or subtracted from particularith coefficient value depending on the sign of the product of En withY(n-i). In this case the rate of adaption is determined by the recursion(clock) rate and the magnitude of lambda λ.

An excitation signal such as Se that drives the adaptive process isreadily implemented as a pseudorandom maximal-length sequence which hasvalues of plus and minus one (1), with a zero mean, and is very widebandwith a flat spectrum. Illustratively, this signal Se is produced by a15-bit shift register with a two-input exclusive-OR gate fed fromarbitrarily selected bit positions (e.g., 1 and 3) on the shiftregister. The output of the exclusive-OR gate is connected to the inputof the shift register. The shift register generates a pseudorandomsignal that repeats only after 2¹⁵ -1 cycles or bits. The excitationsignal Se can be provided at a low level so that it is below thresholdfor users with severe hearing impairment or near threshold in which caseit is perceived as a low-level sound, such as a white noise when noiseis used.

In one embodiment of FIG. 4 a clock rate of 12.5 kilohertz (twice theNyquist rate of at least the frequency of the highest audio component tobe sampled) is derived from circuit 81 of FIG. 3. Adaptive filter 113 isa digital finite impulse response (FIR) filter with a register length of20 stages and preferably even 40 stages. Sound in air takes about 30microseconds to travel one centimeter. A clock rate of 12.5 kilohertz is80 microseconds per repetition period and thus 80 microseconds perfilter 113 register stage. This means that each register stage is asnapshot of the feedback (mixed up with the external input to be sure)each 80 microseconds, and thus about every 2 or 3 centimeters ofacoustic delay in the feedback path is mapped into the shift register inaddition to the delay through receiver 17. Consequently, an acousticfeedback path roundtrip length of illustratively as much as 54centimeters and more is readily accommodated with 20-stage FIR adaptivefeedback filtering in this way. Since reflections from substantiallygreater distances are likely to be low in amplitude in hearing aidapplications, such long-distance reflections are unlikely to materiallyexacerbate a feedback condition. Adaptation to obtain a new set of Hecoefficients when the feedback path changes is preferably accomplishedill less than about 2 seconds and even more preferably in less than asecond.

The system of FIG. 4 and the adaptive process or method also representedby the above equation and description were implemented and evaluated ona laboratory-based digital signal processing system and on anexperimental wearable digital hearing aid. In both implementations anear module consisting of a microphone and receiver as in FIG. 2 in anin-the-ear module and a second version consisting of a microphone andreceiver in a behind-the-ear module coupled acoustically to the ear wereused. In both instances the hearing aid gain Hs could be adjusted sothat oscillation due to acoustic feedback between receiver andmicrophone occurred in the absence of equalization or offsetting. Whenthe offsetting process was used under the same conditions by introducingadaptive feedback filtering as disclosed herein, oscillation ceased.

The performance of the adaptive system is excellent. The apparatus andprocess are stable over a wide range of system gains, system transfercharacteristics and adaption rates. The adaptive system does not requirethat the system process Hs be linear, and indeed in FIG. 5 a systemprocess Hs is shown that includes nonlinear limiter elements. Theadaptive filter 113 coefficients can be updated singly upon theoccurrence of each latest sample, or they can be updated as a whole ortogether essentially simultaneously after a period of time when manysamples have arrived. Mixtures of these two processes are also feasible.The updating of individual coefficients can occur by independentprocesses. Accordingly, numerous embodiments of the invention are madeavailable to the skilled worker.

Any self oscillation which occurs when the undesirable, externalfeedback characteristic Hf changes suddenly, if such oscillation occursat all, is accommodated by the apparatus and processes disclosed herein.The self oscillation signal would act and does act as a distinct signalfrom the sound which is to be amplified by the hearing aid, and is thustreated in similar fashion to already present distinct signal Se.Therefore, when any accidental oscillations occur briefly, if at all,the system continues to adapt to an equalized state of complete offsetof feedback, and no auxiliary circuits and processing to detect andinhibit self oscillation are needed.

The system of FIG. 4 not only prevents self-oscillation, but alsoequalizes or adjusts the system to prevent underdamped ornear-oscillatory conditions from occurring. In conventional hearingaids, near oscillatory conditions result in undesirable signaldistortion and poor performance.

In FIG. 5 a preferred version of filter 109 of FIG. 4 includes foursections of all-hardware filter-limit-filter hardware blocks operatingin four different frequency bands to act as a four-channelfilter-limit-filter digital filter. As in FIG. 4 the combining circuit107 adds an adaptive output from block 113 of FIG. 4 to the digitizedoutput of the external microphone 13. Combining circuit 107 feeds eachof four finite impulse response (FIR) digital filters 131.1, 131.2,131.3 and 131.4. Each of these four filters is provided withcoefficients from the serial interface 45 of FIG. 3 which are stored inRAM 49. The outputs of the four filters 131.1-.4 are respectively fed tofour limiter circuits 133.1, 133.2, 133.3 and 133.4. The limiters arenonlinear devices which prevent or limit an output from the previousfilter blocks from exceeding a predetermined level which also isprovided as input data from the serial interface 45 and stored in RAM49. The outputs of the limiter circuits are in turn respectively fed toadditional digital filters 135.1, 135.2, 135.3 and 135.4. Both thefilters in the group 131 and filters in the group 135 are suitablyfinite impulse response (FIR) filters which are tuned according to thecoefficients provided to them to establish four contiguous frequencybands generally spanning the audible frequency range. The outputs of thefilters 131.1-.4 are all fed to the combining circuit 111 to which theW1-weighted pseudorandom noise Se is also provided to produce the outputY.

FIG. 6 illustrates principles of operation of finite impulse response(FIR) filters generally and of each and any of the finite impulseresponse digital filters 113, 131.1-.4, and 135.1-.4 of FIGS. 4 and 5for instance. An impulse response 141 of a filter is the output of thefilter when it is excited with a single impulse input (as expressed by amathematical delta function). The actual input 143 of a real filter in areal application is of course some general voltage over time and not animpulse. However, the actual voltage input over time can be regarded asa succession of impulses. The impulse responses of the filter to thesuccession of input impulses are curves like impulse response 141,except time-shifted and scaled relative to impulse response 141. Theseimpulse responses are added together in order to determine an output 145of the filter in response to the actual voltage input over time. Theshifting, scaling and adding process is called convolution of the input143 with impulse response 141.

In FIG. 6 the impulse response 141 illustrated for one example, beginsat zero, rises to a peak and then gradually decays away over infinitetime. A set of coefficients C₀, C₁, C₂, C₃, C₄, C₅, ..., C_(i), ...C_(M) of the filter are loaded as digital representations into the RAM49. The coefficients represent selected values on impulse response 141over a finite initial length of time, hence the name "finite impulseresponse" filter.

The input 143 is a series of sample values in digital form which areinput to a shift register for holding the digital values beginning witha most recent sample S_(n) and going backwards in time to previoussamples S_(n-1), S_(n-2), S_(n-3), S_(n-4), S_(n-5), . . . S_(n-i), . .. S_(n-M). Multiplying arithmetic logic circuitry and adding circuitryforms the products C₀ S_(n) plus C₁ S_(n-1) plus . . . plus C_(i)S_(n-i) . . . plus C_(M) S_(n-M). The just-described sum of products atany given time constitutes the electrical output of the FIR filter atthat time. Upon the arrival of an additional input sample all of thesamples in the shift register are moved one to the right so that S_(n)is now the latest most recent sample and S_(n-1) is the sample which wasS_(n) in the previous computation. Then the sum of products isrecalculated forming the next successive electrical output of the FIRfilter. This process is repeated over time to produce a time-domainoutput 145 of the FIR filter as shown in FIG. 6.

For brevity FIR filters where used in the present work are simplyindicated with blocks, it being understood that the operations andcircuitry are provided as appropriate to achieve the filter operationdesired. Coefficients are designated for general discussion herein bythe letter C with a subscript for the particular coefficient.Coefficients for particular filters are also designated by theirtransfer function designator followed by parenthesis designating theparticular coefficient, e.g. He(M). Coefficients for the filters arevaried by associated logic circuitry controlling them, as indicated by aslanting arrow through each filter box. Where no logic circuitry isindicated, and a slanting arrow is used, the logic circuitry is includedwith the filter and the logic circuitry has two inputs, the first inputbeing from the source of the slanted arrow and the second input beingfrom the input to the filter box.

FIG. 7 is an illustration for a heuristic explanation of the adaptivefilter operations which are implemented in a preferred embodiment suchas that of FIG. 4. Most of the elements of FIG. 4 are suppressed forclarity. The pseudorandom noise signal Se is supplied to the hearing aidsystem as bipolar pulses, and an individual pulse of this pseudorandomnoise is shown entering the hearing aid system as a pulse 151.Measurements on one receiver indicate that it has a linear phase delayof about 480 microseconds, meaning that it produces a soundcorresponding to an electrical signal input to it about 480 microsecondslater, or 6 time units of 80 microseconds per sample. A soundcorresponding to the noise signal Se is included in the sound emitted bythe receiver. Some of the sound corresponding to the pseudorandomsignal, also referred to as a probe signal, returns to the microphoneafter a delay period roughly equal to the sum of the 6 time units ofdelay of the receiver plus more delay attributable to the feedback path.

In FIG. 7, the single illustrated pulse 151 passes back to the inputmicrophone and then to combining circuit 107 as part of the feedbackcontribution 44. It will be understood as shown in FIGS. 1 and 8 thatthere are actually many acoustic feedback paths of differing propagationdelay times among which the impulse 151 becomes divided. The variouspreferred embodiments remarkably are able to accommodate the verysubstantial complexity of the many feedback paths. Only three of thefeedback paths are shown in FIG. 8 having delays of eight, nine and tentime units, (including 6 units for receiver 17 delay) for instance.Since the coefficients in adaptive filter 113 of FIG. 4 areindependently updated, it is sufficient for example to describe theinventive operations in logic circuit 117 with respect to a FIG. 8portion of the feedback path with 9 time units of delay.

Referring to FIG. 7, a shift register in the logic circuit 117 of FIG. 4stores the last 20 samples of the pseudorandom noise signal Se (or somesignal such as Y that includes Se). Also, the output of combiningcircuit 107 is fed as combined signal input En to the logic circuit 117.The polarity of combined signal input En is compared with the polarityof each one of the values of the pseudorandom noise signal in the shiftregister store. The ninth stage in the shift register has a polarity ofthe pseudorandom noise signal that matches the polarity of the arrivingpulse 151 because pulse 151 has been delayed for 9 time units. Thecomparing process then produces a positive or increasing magnitude ofoutput for the ninth stage which then is used to successively increasethe magnitude of the ninth coefficient C₉ in the coefficients of FIRdigital filter 113.

FIG. 9 illustrates the comparing process. A first graph 161 shows aprogression of polarities of the input signal En. Another waveform 163represents the polarity of the ninth stage of the shift register of FIG.7 over time. The polarities of waveform 163 exactly match those of thewaveform 161 at any given time. By multiplication or by an exclusive-OR(or exclusive-NOR) operation, the product of the two waveforms 161 and163 over time is always positive as represented by incrementing signalINC in FIG. 9. This consistently positive value of incrementing signalINC is used to persistently increase the magnitude of coefficient C₉.

Referring again to FIG. 7 the sign of the ninth coefficient is arrangedto offset the feedback in combining circuit 107. Since circuit 107 is asummer, the sign of the coefficient is made negative as it increases inmagnitude.

Accordingly, coefficient C₉ is initially increased to a small value 153in FIG. 7 since uncancelled feedback is detected in a feedback path withdelay of 9 time units. The signal Y in FIG. 4 includes the pseudorandomnoise and is filtered by adaptive filter 113 according to coefficientC₉, among other coefficients. That part of the output of filter 113which has a delay of nine time units due to the action of thecoefficient C₉ partially offsets the feedback in the part of thefeedback path with delay of 9 time units. Since the small value 153 forcoefficient C₉ is insufficient to provide enough offset to cancel thecorresponding feedback, then there will still be uncancelled feedbackdetected when the polarity of signal En is next compared with thepolarity in the ninth stage of the shift register in the logic circuit117. As time goes on, coefficient C₉ consequently becomes increased moreand more until it reaches a larger value 155 which successfully cancelsthe feedback in the path with a delay of nine time units. Once theoffsetting is complete, the comparison process in logic circuit 117produces an output of zero and the coefficient C₉ in the filter 113reaches an equilibrium. If the process overshoots or the acousticfeedback decreases due to activities of the hearing aid user, themagnitude of coefficient C₉ is decreased back to equilibrium because thepolarity of the feedback becomes consistently opposite to the polarityof the value in the ninth cell of the shift register of logic circuit117 and the comparing process causes the decrease.

External sound and feedback from feedback paths with other amounts ofdelay in FIG. 8 average out to zero in the comparison process andproduce only statistical fluctuations rather than bias in coefficientC₉. FIG. 10 illustrates this averaging process between a feedbackcomponent of delay 7 and the polarity in shift register stage 9. In FIG.10, waveform 163 (polarity of Se in shift register stage 9) and awaveform 165 (polarity of feedback component with delay 7) are compared.The result, unlike the incrementing waveform INC of FIG. 9, is awaveform 167 which is negative about as much of the time as it ispositive. Therefore, the coefficient C₉, for example, fluctuatessomewhat but on average is unaffected by the signal resulting from thefeedback path portion having delay equal to 7 or any other delay besides9 time units.

Reference is made again to FIG. 7. Since all of the other coefficientsas well as C₉ in the filter 113 are concurrently varied toward theirequilibria by analogous processing, the logic circuit 117 in effectcauses adaptive filter 113 to simulate or mimic the impulse response ofthe entire feedback path to the extent of available coefficients, taps,or length, in the FIR filter 113. Moreover, since all the coefficientsare varied toward equilibrium values which offset and cancel the probesignal returning by acoustic feedback, they are of necessity the correctvalues to offset and cancel the feedback of the external sound itself,thus eliminating the entire feedback contribution in the hearing aid.

As thus described, the combined signal input (e.g. En) is a digitalsignal that has a magnitude and includes a polarity signal representinga changing polarity of the combined signal input. The logic circuit 117has a shift register which acts as an example of a means for temporarilystoring a series of values, connected to the first means (e.g. connectedto Hs 109 and summer 111 of FIG. 4) so that the series of valuesrepresent samples of polarity of the filtered signal combined with thesecond distinct signal, or alternatively represent samples of polarityof the second distinct signal alone. The shift register is also regardedas an example of a means for electronically deriving a series of digitalvalues having polarities responsive to the probe signal. The logiccircuit also has add/subtract circuitry that increases and decreaseseach coefficient in magnitude for the all-hardware digital filter. Theincreasing and decreasing of each coefficient respectively depend onwhether a corresponding value in the series of values has the same oropposite polarity compared to the polarity currently represented by thepolarity signal. As a result, the He filter 113 acts as an example of adigital filter for electronic processing of the probe signal inaccordance with a series of digital coefficients to produce a filteroutput that substantially offsets the feedback contribution in theelectrical output of the microphone in the hearing aid. Remarkably, thepreferred embodiments constantly simulate the impulse response of thefeedback path as it is at any given time and modify the impulse responseas the actual impulse response of the feedback path itself changesduring the daily routine of the user, whereby acoustic feedback isoffset and eliminated.

To even further illustrate advantageous operation in the presence ofexternal sound such as speech, FIG. 11 provides a graph of an inputwaveform of speech 171 (dashed), together with a graph (beneath) ofpseudorandom noise 173. Adding these two waveforms 171 and 173 togetherand detecting the polarity of the sum produces a waveform 175. The noise173 contributes to the polarity of waveform 175 near the zero crossingsof the speech waveform 171. Other periods of positive or negativepolarity due to the speech cause averaging to zero in the comparingprocess with noise 173 in logic circuit 117.

FIG. 12 shows a detailed circuit diagram of logic circuit 117 of FIG. 4.Output Y from combining circuit 111 of FIG. 4 is shifted through astorage circuit such as a series of registers 181.0, 181.1, . . . and181.M which are the counterparts of the stages of the shift register ofFIG. 6 marked with samples S_(n), S_(n-1) . . . S_(n-M). Polaritysignals from each of the registers in the group 181 are respectivelysupplied to a set of exclusive-OR gates 183.0, 183.1 . . . 183.M. Theregisters in the group 181 are efficiently shared with FIR filter 113since they hold the samples of the output Y needed for filter 113. Apolarity bit from the input signal E_(n) is provided in common to anadditional input of every one of the exclusive-OR gates in the group183. The exclusive-OR gates in the group 183 have outputs which controladd-subtract circuits 185.0, 185.1 . . . 185.M. When the polarity of theinput signal E_(n) is the same as the polarity of the contents of anygiven register, such as 181.1 for example, then the exclusive-OR gate183.1 produces a low output. Add-subtract circuit 185.1 correspondingthereto is responsive to an input high to add and to an input low tosubtract. Accordingly, when the polarity of the input signal E_(n) isthe same as the polarity of the contents of register 181.1, add-subtractcircuit 185.1 performs a subtract operation. In the embodiment of FIG.12, a subtract operation decrements by one as indicated by input from asquare block marked "1" for each of the add-subtract circuits in thegroup 185. In this particular embodiment the increasing and decreasingof each coefficient thus occurs by a predetermined amount (e.g.,lambda=1) independent of both the magnitude of the combined signal inputand the magnitude of the filtered signal combined with the seconddistinct signal (e.g. output Y). Repeated subtractions produce anegative coefficient value He(1) for the FIR filter 113 for offsettingpurposes.

A clock output from clock and decoder circuit 81 of FIG. 3 clocks theregisters in the group 181 as well as an additional group of registers187.0, 187.1 . . . 187.M respectively fed by the add-subtract circuits185.0, 185.1 . . . 185.M. The outputs of the registers in the group 187are respectively coupled to an additional input of the correspondingadd-subtract circuits in the group 185. As a result of the operations ofthe logic circuit 117 of FIG. 12, the registers in the group 187 produceoutputs which digitally represent the coefficients He(O), He(1), . . .He(M) for the adaptive filter 113.

If the entire register length of each register 187.0, 187.1 . . . 187.Mis used to represent the value of each filter coefficient, then a degreeof statistical fluctuation can occur in each coefficient in adaptiveoperations. An uncomplicated approach for reducing statisticalfluctuations in coefficients is to connect higher-order bits of eachcoefficient from each register to FIR filter 113, leaving the two (orother selected number) lowest order bits (or least significant bits)isolated from FIR filter 113.

The circuitry of FIG. 12 is thus an example of a logic circuit foradaptively varying the digital coefficients of an all-hardware digitalfilter as a function of polarity of the combined signal inputindependently of its magnitude so that the adaptive output substantiallyoffsets the feedback contribution in the electrical output of themicrophone.

FIG. 13 illustrates circuitry by which the polarity signal En(polarity)of FIG. 12 is derived. The output of combining circuit 107 of FIG. 4 issuitably an 8-line data bus 201 for carrying signal En. In FIG. 13 oneof the lines of the data bus 201 represents the polarity or positive ornegative character of the input signal En. The rest of the linescollectively carry a parallel digital signal which represents themagnitude of the input signal En.

FIG. 14 illustrates that in other embodiments the input signal En isanalog. The analog signal is fed to a comparator 203 which acts as apolarity detector and produces a digital 1 or 0 corresponding to thepositive or negative polarity of the input signal En.

Other embodiments for even further improved operation are nextdiscussed. It is desirable to reduce any uncertainty or statisticalfluctuations in estimating the filter coefficients that represent asimulation of the external feedback path. For instance, the externalsound signal Xi and its amplified counterpart generated by the receiver17 are believed to introduce some fluctuations, see discussion inconnection with FIG. 11. These fluctuations can be reduced by the properchoice of the probe signal level and the combined signal delay of thehearing aid and the external feedback path.

Therefore, improvements of apparatus and method such as: 1) maintaininga constant ratio between the external signal level and the internalexcitation signal used to "probe" the feedback path, and 2) to addprocessing delay so that the autocorrelation of the external signal inthe error function (combined signal input En) is small, can be effectivein reducing the uncertainty of estimation.

A further measure 3) involves supplying the probe signal such as noiseSe, instead of output Y, to logic circuit 117. Since the external signalXi and the internal probe signal Se are uncorrelated, it is possible toseparate the contributions of the external signal Xi and internal probesignal Se to the combined signal input En by directly comparing thesignal En with the probe signal Se itself, and thereby advantageouslyaccomplish an operation having the same benefits as cross correlatingthe signal En with the probe signal itself. This can be shown asfollows. The signal En in FIG. 4 is given by a sum of convolutions

    En=g1*Xi+g2*Se                                             (3)

where En, Xi and Se are the combined input signal (also called the errorsignal herein), external signal, and internal probe signal respectivelyand g1 and g2 are the inverse transforms of a pair of transfer functionsG1 and G2 respectively given by

    G1=Hm/(1-HsH) and                                          (4)

    G2=H/(1-HsH)                                               (5)

where H=HrHfHm+He is the combined transfer function of the externalfeedback path and the internal equalization feedback path and Hs, Hr andHm are as shown in FIG. 4.

An even more elaborate analysis establishes G1 and G2 by recognizingthat

    (Xi+HfHr(HsEn+Se))Hm+He(HsEn+Se)=En                        (6)

Rearranging yields

    XiHm+(HfHrHm+He)Se=En(1-HeHs-HfHrHsHm)                     (7)

Dividing through by the parenthesized quantity after En and comparingwith the hereinabove defining equation for G1 and G2 yields

    G1=Hm/(1-HeHs-HfHrHsHm)                                    (8)

    G2=(HfHrHm+He)/(1-HeHs-HfHrHsHm)                           (9)

A cross-correlation of the error signal En with the probe signal Seproduces

    E[En(n)·Se(n-i)]=g2(i)                            (10)

because the probe signal is uncorrelated with the input signal Xi andthe term involving g1 advantageously vanishes.

Accordingly, g2(i) thus obtained is an unbiased estimator of thedifference between the external feedback path and the internalequalization path. Therefore, g2(i) can be used to adaptively adjust thecoefficients of the adaptive filter or equalization filter to cancel theexternal feedback path according to an adjustment process defined by

    He(j+l)=He(j)-g2(j)                                        (11)

where He refers to the vector of coefficients (and not to the transferfunction itself in the Laplace or z domain) and j is an iteration indexin the adjustment process and not a designator (like index i) of aparticular coefficient. g2(j) refers to the vector of cross-correlationsat iteration j.

When g2 is reduced to the zero vector, the adaptive filter transferfunction He in this example cancels the feedback path according to theequation (involving the transfer functions themselves and not theirinverse transforms)

    He=-HrHfHm                                                 (12)

"Equalization" as the term is used herein, refers to the process ofmaking the coefficients of the adaptive filter process simulate thecoefficients which would describe the path HrHfHm so that feedback canbe offset by subtraction, summation or any other combining process, andis not to be confused with equalization as the term is used in someother contexts in the literature for series filtering of communicationchannels, for instance.

These processes can be implemented by a variety of apparatusembodiments. In one embodiment, an electronic block computation of crosscorrelation is performed over one period of the probe signal which is amaximal-length noise sequence. The resulting values of g2(j) are thensubtracted from the coefficients corresponding to He(j) to update theadaptive filter.

FIG. 15 shows a further VLSI embodiment incorporating improvements toreduce fluctuations caused by the external sound in the estimation ofthe feedback path. Only those circuit portions which are different fromthe circuitry of FIG. 4 are explained in additional detail. The outputof system process digital filter 109 is fed to an additional combiningcircuit 211 and weighted with a weighting factor W2. Weight W2 is set tosome fraction in a range, illustratively, of about 0.1 to 0.5.Pseudorandom noise signal Se is an additional input to the combiningcircuit 211. At normal levels of speech the weighted contribution of theoutput of filter 109 does not affect the polarity of its sum in circuit211 with noise signal Se (see FIG. 18 and discussion later hereinbelow).The output of the combining circuit 211 is fed as a signal U to thelogic circuit 117.

Logic circuit 117 operates in generally the same way as describedhereinabove in connection with FIGS. 4, 7 and 12. However, in anabnormal situation in hearing aid use, a feedback instability mightsubstantially increase the output of filter 109. The output would thenexceed normal speech loudness levels and contribute in apolarity-determinative manner to the output U of combining circuit 211notwithstanding the fractional value of W2. The input En and the outputU would correlate substantially to one another due to the feedbackitself. Combining circuit 211 thus causes logic circuitry 117 to speedup the adaption process in the abnormal situation, using the feedbackitself for controlling the logic circuitry 117 to eliminate theinstability.

In another improvement, noise input weight W1 of combining circuit 111of FIG. 15 is made variable by adding an automatic gain control (AGC)circuit 215. AGC circuit 215 is responsive to output S of system filter109. AGC circuit 215 is made to operate so that the amount ofpseudorandom noise signal Se which is combined with the output S offilter 109 increases directly with average sound volume level in theoutput S. In this way, when the external sound is faint, there is lesspseudorandom noise injected into the hearing aid circuit.Advantageously, the presence of the pseudorandom noise signal becomesless noticeable at such times due to the action of automatic gaincontrol circuit 215.

On the other hand, when sound output S of filter 109 has greatermagnitude due to loud external sounds, a substantial amount ofpseudorandom noise signal is injected into the hearing aid circuit atcombining circuit 111 due to an increased value of weight W1 establishedby the AGC circuit 215 in response to the output S. In this way, asignificant presence of the pseudorandom noise signal is established atthe zero crossings of speech (compare FIG. 11) at all speech magnitudes,and the operation of logic circuit 117 is made even more reliable andeffective in controlling the adaptive filter 113.

ACG circuit 215 suitably has an energy or envelope detector (such as arectifier and low pass filter) connected to a voltage controlledmultiplier connected to the Se noise generator to vary the noise levelprovided to the input of combining circuit 111 as described herein.

AGC circuit 215 thus is an example of a means for varying the seconddistinct signal (e.g. Se) in magnitude as a function of the magnitude ofthe filtered signal, and indeed as a direct function of an average ofthe magnitude of the filtered signal. Also, the circuitry includingcombining circuits 111 and 211 constitutes an example of first combiningmeans for combining the filtered signal with a second distinct signalfor the receiver of the hearing aid in a proportion wherein themagnitude of the second distinct signal is generally less than themagnitude of the filtered signal at loudness of normal speech and secondcombining means for combining the filtered signal with the seconddistinct signal to produce a control signal or second combined signalfor affecting the processing by the means for electronic processing(e.g. filter 113) wherein the magnitude of the second distinct signal isgenerally greater than the magnitude of the filtered signal at loudnessof normal speech. The logic circuit 117 of FIG. 15 adaptively varies thefilter coefficients He(0), He(1), . . . He(M) of filter 113 as afunction of polarity of the second combined signal independently of itsmagnitude.

FIG. 16 shows an alternative circuit arrangement for logic circuit 117of FIGS. 4 and 15. Corresponding circuitry is numbered withcorresponding numerals in FIG. 16. Add-subtract circuits in the group185 are supplied with inputs that are produced by a set of multiplyingcircuits 221.0, 221.1 . . . 212.n which have one input respectively forone or more of the magnitude bits from the corresponding registers inthe group 181. The shift register group 181 derives its input eitherfrom the input Y of FIG. 4 or the input Y or U of FIG. 15 as the skilledworker elects. An additional input to each of the multiplying circuits221.0, 221.1, . . . 221.n is derived from the most significant bit (MSB)of the combined input signal En.

In this way, the add-subtract circuits in the group 185 increase ordecrease their corresponding coefficients in the registers 187, namely,He(O), He(1), . . . and He(M). A greater accuracy of estimation isobtained by the circuit of FIG. 16 at the expense of increasedcomplexity of providing the multiplying circuits.

FIG. 17 illustrates another all-hardware VLSI processing apparatus 231according to another preferred embodiment in which a VLSI die 241 has aprocessing area 243. Processing area 243 includes a circuit 245 forimplementing system process Hs filter functions which is fed by theoutput of a microphone Hm through a combining circuit 247. The output ofthe combining circuit is also fed to a logic circuit for adaptivelycontrolling an additional filter section He numbered 249. The output ofthis filter 249 is fed back to combining means 247. Similarly, an outputS of filter 245 is supplied to first and second combining circuits 251and 253. A pseudorandom noise signal generator 255 also feeds combiningcircuits 251 and 253 respectively. The noise is weighted with weight W1in combining circuit 251 and the Hs filter output S is weighted with aweight W2 at the input of combining circuit 253. An output Y ofcombining circuit 251 is supplied to block Hr and also is fed back tofilter 249. The LOGIC control block for filter 249 is fed with signal Ufrom combining circuit 253.

In this way, the processing area 243 and its circuitry constitutes anexample of a means for electronic processing of the electrical output ofthe microphone means to produce a filtered signal. The combining circuit251 acts as an example of a means for combining a filtered signal with asecond distinct signal for the receiver of a hearing aid in a proportionwherein the magnitude of the second distinct signal is generally lessthan the magnitude of the filtered signal at loudness of normal speech.The combining circuit 253 acts as an example of a means for combiningthe filtered signal with the same second distinct signal to produce acontrol signal for the means for electronic processing wherein themagnitude of the second distinct signal is generally greater than themagnitude of the filtered signal at loudness of normal speech.

FIG. 18 shows the relative proportion of normal speech 261 when weightedin accordance with the weight W2 in circuit 211 of FIG. 15 or circuit253 of FIG. 17. The level of the speech 261 is less than about a halfthe level of pseudorandom noise 263 of FIG. 18.

FIG. 19 illustrates the relative proportion of normal speech 271 whenpseudorandom noise 273 is combined with weight W1 in circuit 251 of FIG.17 or circuit 111 of FIG. 15. When the normal speech is at a loud volumethe weight W1 is adjusted so that the height of the pseudorandom noiseis on the order of one tenth to one half of the amplitude of the speech.However, in FIG. 20 speech 277 at lower loudness levels is also used tocontrol AGC circuit 215 to deemphasize or reduce the level of the noisepulses 275 so that they bear approximately the same proportion to theamplitude of the lower volume speech 277 as noise 273 bears to louderspeech 271 of FIG. 19.

FIG. 21 illustrates a circuit in which a combining circuit 281 isimplemented as a subtracter instead of a summer at the input of a systemprocess Hs filter 283. In FIG. 21 adaptive FIR filter 113 is driven by alogic circuit 285 which is appropriately modified to take account of thechange to a subtracter. The Hs filter 283 suitably has sections of delayadded to it in order to even further reduce the likelihood of feedbacksquealing and instability. Logic circuit 285 acts as an example of ameans for temporarily storing a series of values, which temporarystoring means is connected to said first means (281, 283, 111) so thatthe series of values represent samples of polarity of the seconddistinct signal (e.g. Se), and means for increasing and decreasing eachcoefficient in magnitude for all-hardware digital filter means by afunction of the combined signal input, the increasing and decreasing ofeach coefficient respectively depending on whether a corresponding valuein the series of values has the same or opposite polarity compared to apolarity signal representing the polarity of the combined signal input.

In FIG. 22 logic circuitry 285 of FIG. 21 is detailed. The logiccircuitry 285 of FIG. 22 is analogous to the circuitry of FIG. 12 exceptthat a set of add/subtract units 285.0, 285.1, . . . 285.M aresubstituted for the units in the group 185 of FIG. 12. As indicated bycontrol input "bubbles," the add/subtract units in the group 285 of FIG.22 have their control input low active for addition instead ofsubtraction, and high active for subtraction instead of addition. Thismeans that an add/subtract unit in FIG. 22 adds when the output of itscontrolling exclusive OR gate is low instead of when it is high as inFIG. 12. The result is that the coefficients of the adaptive filter 113of FIG. 21 have the opposite sign from the coefficients which aregenerated by logic circuit 117 of FIG. 12 for the adaptive filter 113 ofFIG. 4.

The output of the FIR filter 113 of FIG. 21 is thus made of theappropriate sign to operate the subtracting circuit 281 of FIG. 21 in amanner to effectively offset the feedback contribution. By way ofcomparison, in FIG. 4 the output of the FIR filter 113 is of the propersign to cooperate with a summing circuit used as the combining circuit107 of FIG. 4.

Circuitry 285 in FIG. 22 constitutes an example of a means for adaptingonly in response to polarities of signals supplied to and from firstfilter means (e.g., filter 283 of FIG. 21). Each of the exclusive-ORgates 183.0, 183.1, . . . 183.n is supplied with a single line carryinga digital signal representative of the polarity of the combined inputsignal En. A series of D latches 291.0, 291.1, . . . 291.M are connectedas a shift register to hold digital signals respective of the polarityhistory of any appropriate signal with which signal En is to becompared. An appropriate signal is any of noise signal Se, or signal Uof FIG. 17 or signal Y of FIGS. 4, 17 or 21, for example. The signalutilized is supplied to the first D-latch or bit register 291.0 and thenthe successive polarities of this signal are shifted into the rest ofthe registers 291.1 to 291.n. A signal from the clock circuit 81 of FIG.3 is connected to clock the registers in the group 187 and the D-latchesin the group 291 in FIG. 22. In this way, the coefficients of theimpulse response for adaptive filter 113 are successively increased anddecreased according to the principles discussed in connection with FIGS.6, 7, 9 and 10.

FIG. 23 shows an embodiment for reducing statistical fluctuations bykeeping running totals in a first set of registers 301.0, 301.1, . . .301.M and updating the coefficients He(0), He(1), . . . He(M) lessfrequently in a second set of registers 303.0, 303.1, . . . 303.M. A setof add/subtract circuits 305.0, 305.1, . . . 305.M increase and decreaseeach running total in the first set of registers 301.0, 301.1, 301.M bythe value of combined signal input En, including sign and magnitude,depending on whether a corresponding value in the series of digitalvalues such as polarity of Se, U or Y in the shift register 291.0,291.1, . . . 291.M have the same or opposite polarity compared tocombined signal input En.

In the embodiment of FIG. 23 the exclusive-OR gates in the group 183 ofFIG. 22 are omitted. Instead combined signal input En is itself suppliedto every one of the add/subtract circuits in the group 305. In this waythe add/subtract circuits are an example of means responsive to thetemporarily stored series of digital values representing polarities ofthe pseudorandom noise signal for increasing or decreasing each runningtotal in magnitude by an amount directly related to the magnitude of thecombined signal input.

The stages of the shift register 291 are supplied to low active Addinputs of the add/subtract circuits 305 respectively. A clock circuit 81provides clock pulses for actuating all of the stages in the shiftregister 291 as well as all of the registers in the group 301. Theoutput of every register in the group 301 is connected back to itscorresponding add/subtract circuit 305.0, 305.1, . . . 305.Mrespectively so that running totals r(0), r(1), . . . r(M) have thevalue of signal En added to or subtracted from them.

Running totals r(0), r(1), . . . r(M) in the registers in the group 301as a first set of registers are respectively supplied to correspondingadding logic circuits 307.0, 307.1, . . . 307.M. Each of the second setof registers in the group 303 for holding coefficients has its outputconnected to the corresponding input of the add circuits in the group307. A divide-by-L circuit 309 has an input connected to the clockcircuit 81 to divide the clock pulses by a constant L such as 20.Circuit 309 initiates updating of registers 303 by adding the runningtotals respectively to the coefficients, and then resets the runningtotals to zero in the registers in the group 301 at intervals.

In this way, the divide by L circuit 309 together with the add/subtractcircuits in the group 307 constitute an example of a means forrespectively adding the running totals in the first set of registers(e.g., group 301) to the coefficients in the second set of registers(e.g., the group 303) to update the coefficients for the digital filter(e.g., filter 113) less frequently than the increasing and decreasing ofthe first set of registers occurs. Circuit 309 is an example of a meansfor frequency dividing the clock pulses to supply a second set of clockpulses to the second set of registers to update them at a frequencydivided rate from the means for adding. Clock circuit 81 is an exampleof a means for supplying clock pulses to initiate the incrementing anddecrementing of the first set of registers.

This type of operation advantageously averages out statisticalfluctuations in the running totals as they are developed. The operationthereby produces a more effective offsetting of the feedbackcontribution and decreases the frequency at which the coefficients areupdated. The statistical fluctuations are believed to be reduced by afactor on the order of the reciprocal of the square root of L, which isa percentage reduction of over 75% when L=20.

In FIG. 24 a polarity-only version of the circuitry of FIG. 23 isillustrated. The circuitry is the same as FIG. 23 and numerals areprovided identically to corresponding elements. Only the polaritycomponent of the input signal En is connected to a series ofexclusive-OR gates 311.0, 311.1, . . . 311.M. The outputs of theexclusive-OR gates respectively feed low active Add inputs ofadd/subtract circuits in the group 305. The add/subtract circuits aceuncomplicated up/down counter implementations with low activeincrement/decrement inputs used as the equivalent of add/subtractinputs. The running totals are incremented and decremented by 1 asindicated by boxes 313.0, 313.1, . . . 313.M labelled "1". In this way,logic circuitry of FIG. 24 includes a first set of registers 301.0-.Mfor holding running totals that are incremented and decremented as afunction of polarity of the combined signal input and polarity of atleast one signal from the first means (e.g. Se, U or Y). The circuitryis relatively uncomplicated and economical, produces running totals andaverages out statistical fluctuations as the running totals aredeveloped with advantageous corresponding effect on the coefficients forthe adaptive filter 113.

In FIG. 25 a graph of frequency response of a hearing aid circuit in dB(decibels) versus frequency in the range 0-5 kilohertz is shown.Measured frequency response of the hearing aid circuit with the adaptivefilter disconnected and therefore unable to offset the feedbackcontribution is shown as solid line with numerous sharp, jagged andprominent peaks 331.1, 331.2, 331.3, 331.4,331.5, 331.6 and 331.7.Without offsetting, the uncorrected frequency spectrum thus exhibitspeaks that are indicative of highly underdamped behavior correspondingto incipient instability and "ringing" in the hearing aid circuit. Whenthe volume of the hearing aid is increased beyond a certain point itstarts to squeal due to the feedback.

Using inventive apparatus and methods to offset the feedbackcontribution, the measured frequency response of the hearing aid circuitis as shown in the dashed portions 333.1 through 333.7. It is clear thatthe peaks when the feedback is offset are reduced and far less prominentand establish a frequency spectrum that is much more even in itsresponse. As a result, volume levels available from circuitry utilizingthe inventive apparatus and methods are substantially increased comparedto those of circuitry lacking the offsetting capabilities. Stable,equalized system behavior is obtained in acoustic conditions of use thatwould have caused oscillation without the adaptive equalization ineffect. Accordingly, the resulting hearing aid output is substantiallyeasier to listen to and understand by the hearing impaired user.

In the present work it has been discovered that the use of anall-hardware digital adaptive filter is very substantially preferable tomicroprocessor-based operations. Table 1 shows the characteristics ofcircuitry which would be involved in the microprocessor estimationapproach as compared with the much simpler, less expensive and much morepreferable arrangement in the preferred embodiments using theall-hardware digital adaptive filter approach.

As shown by Table 1, the all-hardware arithmetic is relatively simplewhile the microprocessor arithmetic unit is relatively elaborate. Themicroprocessor estimation approach inherently involves program addresscircuitry, memory space for a stored program to do the estimation, andinstruction register and decoding hardware, all of which are dispensedwith in the all-hardware digital adaptive filter preferred embodiments.Furthermore, input-output logic and control logic in microprocessorcircuitry on a VLSI die would be much more elaborate than the relativelysimple circuitry which is disclosed herein and is used with theall-hardware digital adaptive filter approach.

                  TABLE 1                                                         ______________________________________                                                            All-Hardware                                              Microprocessor Estimation                                                                         Digital Adaptive Filter                                   ______________________________________                                        Arithmetic   Elaborate  Simpler                                               Unit                                                                          Program      Present    None                                                  Address                                                                       Circuitry                                                                     Memory Space Present    None                                                  for program                                                                   Instruction  Present    None                                                  registers                                                                     and decoding                                                                  Input/Output Elaborate  Simpler                                               logic                                                                         Control      Elaborate  Simpler                                               logic                                                                         ______________________________________                                    

In this way and as shown in FIGS. 4, 15, 17, 21, 26 and 27, for example,there is provided first means fabricated on a VLSI die for electronicprocessing of the electrical output of the microphone to produce afiltered signal and for combining the filtered signal with a seconddistinct signal for the receiver of the hearing aid. All-hardwaredigital adaptive filter means is united with, and connected to, thefirst means on the VLSI die for processing the filtered signal andsecond distinct signal to produce an adaptive output to the first meansto substantially offset the feedback contribution ill the electricaloutput of the microphone in the hearing aid.

The all-hardware digital adaptive filter includes an all-hardwaredigital filter united with, and connected to, the first means forprocessing on the VLSI die for processing the filtered signal and seconddistinct signal in accordance with a series of digital coefficients toproduce the adaptive output. Also provided is circuitry for combiningthe adaptive output with the electrical output and feedback contributionfrom the microphone to produce a combined signal input to the firstmeans for processing. Further, logic circuit means is united with, andconnected to, the first means on the VLSI die for adaptively varying thedigital coefficients of the all-hardware digital filter as a function ofthe combined signal input so that the adaptive output substantiallyoffsets the feedback contribution in the electrical output of themicrophone.

FIG. 26 illustrates a further alternative embodiment of a feedbackoffsetting circuit. This alternative involves an infinite impulseresponse (IIR) filter which can increase the range or effectiveness ofthe offsetting even further with only a modest amount of extra VLSIhardware.

As illustrated in FIG. 26, an FIR adaptive digital filter 401 has itsinput connected to receive the output Y of combining circuit 111. Filter401 is shown as a block which includes a logic circuit 285 of FIG. 21combined with a filter 113 of FIG. 21. The logic circuit in filter 401is responsive to combined signal input En and output Y. An additionalfilter block 403 has both its input and output connected to a combiningcircuit 405. Combining circuit 405 sums the output of filter 403 and theoutput of filter 401 to produce a sum output X that is fed both to theinput of filter 403 and to an input of subtractive combining circuit 281for offsetting feedback present in the output of microphone block 101.

Filter 403 has the same internal construction as filter 401. In otherwords filter block 403 has a logic circuit identical to logic circuit285 of FIG. 21 combined with a filter like 113 of FIG. 21. The logiccircuit in filter 403 is responsive to combined signal input En, thesame as the logic circuit in filter 401. However, the logic circuit infilter 403 is responsive to output signal X, instead of output signal Ythat the logic circuit of filter 401 responds to.

Further, the output of filter 403 is fed back to its own input viacombining circuit 405. Filter 401 has no feedback directly back toitself from combining circuit 405 in FIG. 26. Filters 401 and 403 withcombining circuit 405 together are an example of an IIR filter; however,sometimes filter 403 and circuit 405 alone are also called an IIR filteras context dictates, for purposes of the present work. Thus, the circuitof FIG. 26 represents an improvement over the circuit of FIG. 21, forinstance, by adding an IIR filter in series with the FIR filter of FIG.21.

The FIR adaptive filter is believed to be well suited to compensate fora component of the feedback path Hf 105 that is comprised of acousticdelay and attenuation. The IIR adaptive filter (403, 405) is believed tobe well suited to compensate for resonances that exist in the feedbackpath. Resonances and very long feedback paths in hearing aids, publicaddress systems and other electroacoustic systems are even furtherobviated due to the internal feedback and memory, in effect, that theIIR filter approach provides.

Together, filters 401 and 403 with combining circuit 405 are an exampleof a digital adaptive filter He. This filter He is interconnected with afirst circuit (e.g. blocks 281, 109, 111, 115) and responsive to signalssupplied from the first circuit. The filter He provides electronicinfinite impulse response (IIR) processing of the combination of thefiltered signal (e.g. output of Hs 109) and second distinct signal (e.g.Se 115) so that output X to the first circuit substantially offsets thefeedback contribution in the electrical output of the microphone in thehearing aid. The digital adaptive filter He includes a first digitalfilter 401 and a second digital filter including filter 403 andcombining circuit 405. The first digital filter 401 has an input 407 forsignal Y, which is one example of a filtered signal combined with asecond distinct signal. Filter 401 has an output 409 connected tocombining circuit 405 of the second digital filter. The second digitalfilter produces an output X, part of which is fed back to filter 403 atan input 411 of block 403. Output X of the second digital filter issupplied to the first circuit for electronic processing at subtractingcombining circuit 281 to substantially offset the feedback contributionin the electrical output of the microphone means in the hearing aid.

Updating the filter coefficients in filter blocks 401 and 403 is similarto the processes discussed earlier hereinabove, except that two sets ofcoefficients are involved. Coefficients for filter 401 are designatedw_(k), and coefficients for filter 403 are a_(k), where k is an index(similar to i hereinabove) for designating each particular coefficient.Recursive expressions for updating the coefficients are:

    w.sub.k (n+l)=w.sub.k (n)+sgn[En·Y(n-k)]          (13)

    a.sub.k (n+l)=a.sub.k (n)+sgn[En·X(n-k)]          (14)

In the circuit of FIG. 26, filter block 403 has a transfer function Athat is adaptively varied. The transfer function represented by thesymbol "A" can be a remarkably complicated function in the Laplace (or sor z) domain. Filter block 403 together with combining circuit 405 andfeedback thus provided have a transfer function 1/(1-A). If the transferfunction of FIR filter 401 be represented by W, then the transferfunction He of the entire filter assembly 401, 403, 405 of FIG. 26 isW/(1-A).

In FIG. 27 a further improvement adds an error filtering feature. Filter401 of FIG. 26 is shown as a filter 413 with logic circuit 417 of FIG.27. Logic circuit 417 is responsive to output Y and to a filtered errorsignal V to be described presently. A filter 423 and logic circuit 425of FIG. 27 correspond to filter block 403 of FIG. 26. A combiningcircuit 427 sums the output of the filters 413 and 423 to provide outputX. Output X is also fed back to filter 423.

Logic circuit 425 is shown apart from filter 423 because itadvantageously doubles as the logic circuit for an additional errorfilter 431. Error filter 431 has its input connected to receive thecombined signal input En, which is a form of error signal for purposesof feedback filter He. A combining circuit 433 shown as a subtractor isfed with combined signal input En and also with the output of errorfilter 431. The output V of the circuit having filter 431 and circuit433 is a filtered error signal of the form

    V=En(1-A)                                                  (15)

Logic circuit 425 provides identical coefficients to filters 423 and431. Logic circuit 425 is responsive to outputs V and X and isimplemented with any of the circuits of FIGS. 12, 16, 22, 23 and 24 andcan use polarity-only or all polarity and magnitude information incombined signal input En. A set of recursive expressions for updatingthe coefficients of the three filters 413, 423 and 431 of FIG. 27 are

    w.sub.k (n+l)=w.sub.k (n)+sgn[V·Y(n-k)]           (16)

    a.sub.k (n+l)=a.sub.k (n)+sgn[V·X(n-k)]           (17)

There is no need for three equations because the coefficients a_(k) forboth filter 423 and 431 are correspondingly identical.

Note that in each of FIGS. 26 and 27, a dotted box has been drawn aroundthe digital adaptive filter assemblies corresponding to the He box ofsome earlier block diagrams hereinabove.

The additional error filtering with filter 431 and subtracting circuit433 of FIG. 27 is believed to provide an even further advantageousfeature in smoothing and making even more effective the operation of theadaptive filtering circuitry. A tentative heuristic explanation for theimprovement is believed to be that elements 423 and 427 could form aless than fully stable loop (423,427, 281, 425, 417, 413) due to the1/(1-A) transfer function having a denominator (1-A) approaching zero atsome frequency. By adding elements 431 and 433 having a transferfunction (1-A) (which is the reciprocal of 1/(1-A)), a multiplication ineffect in the Laplace or frequency domain of (1-A)×1/(1-A) occurs sothat such denominator effects are cancelled. It is emphasized that thisexplanation is in no way intended to be limitive or exclusive of otheradvantageous effects and operations which are also present in thisembodiment.

The filter 431 and combining circuit 433 thus provide an example of athird digital filter for filtering a signal (e.g. combined signal inputEn) that is also supplied to first means for electronic processing.Filter 423 is an example of a digital filter section that has a transferfunction and produces an electrical signal which is fed back to itself.Filter 431 with combining circuit 433 provide an example of a means forelectronically filtering the combined signal input according to atransfer function which is substantially the reciprocal of the transferfunction of the digital filter section, to drive logic circuitry. Logiccircuits 425 and 417 together act as a type of logic circuitry fed bythe third digital filter for controlling first and second digitalfilters and thus controlling infinite impulse response (IIR) processing.

Numerous alternative connections to the logic circuits 417 and 425 anduse of multiple combining circuits are also available according to thedescription hereinabove of FIGS. 15, 17 and 21.

A process diagram of FIG. 28 shows illustrative operations of a hearingaid according to some of the inventive methods. Operations commence witha START 501 and proceed to a step 503 to process the electrical outputof the microphone 13 into a combined signal input En and to produce afiltered signal according hearing aid amplification and filtering Hs.

In a next step 505, a probe signal such as a pseudorandom noise signalis generated as an example of a second distinct signal that is distinctfrom the filtered signal just referred to. In a following step 507, thefiltered signal is combined with the second distinct signal for thereceiver of the hearing aid. The second distinct signal is suitablyweighted by a factor W1 so that the magnitude of the second distinctsignal thus weighted is generally less than the magnitude of thefiltered signal at loudness of normal speech.

If the magnitude of the filtered signal has changed, then a step 509varies the magnitude of the second distinct signal (e.g. noise) as afunction of the magnitude of the filtered signal. For instance, in oneembodiment it is desirable to vary the magnitude of noise in directcorrespondence with the magnitude of the filtered signal so that themagnitudes have an approximately constant ratio.

Succeeding step 511 combines the filtered signal with the noise byweighting the filtered signal by a factor W2 so that the magnitude ofthe noise is generally greater than the magnitude of the filtered signalat loudness of normal speech.

Next a step 513 derives a series of digital values having polaritiesresponsive to the probe signal, as by shifting pseudorandom noisethrough a shift register. Then in a step 515 running totals areelectronically maintained in a set of registers. Each running total isincreased and decreased depending on whether a corresponding value inthe series of digital values has the same or opposite polarity comparedto the combined signal input.

A test step 517 tests an index N to determine whether it has reached apredetermined value such as 20. If so, the running totals of step 515ere added to update a set of digital filter coefficients, and index N isreset to 1. If index N has not reached 20, operations branch from step517 to step 521 to increment index N by one, and step 519 is bypassed.In this way, the running totals in the set of registers are respectivelyadded electronically to the coefficients in step 519 less frequentlythan the increasing and decreasing of step 515 occur.

After either step 519 or 521, operations proceed to a step 523 toadaptively filter the filtered signal and second distinct signal (orprobe signal) combined in step 511, for instance. The filtering isaccomplished in accordance with digital coefficients varied only inresponse to polarities, for example, to produce an adaptive output inthe hearing aid. An IIR version of the process in this step 523adaptively filters the filtered signal and second distinct signal toproduce all electrical signal and feeds back the electrical signal asshown by arrow 525 so that the electrical signal is adaptively filteredin the same step 523 also, thereby producing the adaptive output.

In a further step 527, the adaptive output is combined with themicrophone output, substantially offsetting with the adaptive output theacoustic feedback contribution in the electrical output of themicrophone for the processing step 503. Operations loop back to step 503through a test 529 if the process is to continue. If not, operationsbranch from test 529 to an END 531.

The invention comprehends numerous embodiments using digital or analogtechnology and incorporating software, hardware, or firmware asapplications dictate. Applications, combinations and processes forhearing aids, public address systems and other electroacoustic systemsgenerally for use in air, underwater, or in other environments arewithin the scope of the invention.

In view of the above, it will be seen that the several objects of theinvention are achieved and other advantageous results attained.

As various changes could be made in the above constructions withoutdeparting from the scope of the invention, it is intended that allmatter contained in the above description or shown in the accompanyingdrawings shall be interpreted as illustrative and not in a limitingsense.

What is claimed is:
 1. An electronic filter for a hearing aid having amicrophone for generating an electrical output from sounds external to auser of the hearing aid and an electrically driven receiver for emittingsound into the ear of the user of the hearing aid, some of the soundemitted by the receiver returning to the microphone to add a feedbackcontribution to its electrical output, the electronic filtercomprising:an electronic processor for processing the electrical outputof the microphone to produce a first signal and for combining the firstsignal with a second distinct signal to produce a combined first andsecond signal for the receiver of the hearing aid, said second distinctsignal representing noise and being distinct from the first signal; andan adaptive filter, interconnected with the electronic processor forprocessing the combined first and second signal while the electronicprocessor is producing the first signal, said adaptive filter producingan adaptive output to the electronic processor to continuouslysimultaneously offset the feedback contribution in the electrical outputof the microphone in the hearing aid.
 2. An electronic filter as setforth in claim 1 wherein the adaptive filter includes means for adaptingonly in response to polarities of the output of the microphone, and thefirst signal.
 3. An electronic filter as set forth in claim 2 whereinthe adaptive filter comprises an all-hardware digital filter connectedto the electronic processor for processing the combined first and thesecond signal in accordance with a series of digital coefficients toproduce the adaptive output;wherein the electronic processor includes asignal combiner for combining the adaptive output with the electricaloutput and feedback contribution from the microphone to produce acombined signal input to the digital filter; and wherein the adaptivefilter further comprises a logic circuit for adaptively varying thedigital coefficients of the all-hardware digital filter as a function ofthe polarity of the combined signal input independently of its magnitudeso that the adaptive output substantially offsets the feedbackcontribution in the electrical output of the microphone.
 4. A hearingaid comprising:a microphone for generating an electrical output fromsounds external to a user of the hearing aid; an electrically drivenreceiver for emitting sound into the ear of the user of the hearing aid,some of the sound emitted by the receiver returning to said microphoneto add a feedback contribution to its electrical output; an electronicprocessor for processing the electrical output of the microphone toproduce a first signal and for combining the first signal with a seconddistinct signal to produce a combined first and second signal for thereceiver of the hearing aid, said second distinct signal representingnoise and being distinct from the first signal; a controller, receivingthe first signal, for varying the second distinct signal in magnitude asa function of the magnitude of the received first signal; and anadaptive filter, interconnected with the electronic processor, forprocessing the combined first and second signal to produce an adaptiveoutput to the electronic processor to substantially offset the feedbackcontribution in the electrical output of the microphone in the hearingaid.
 5. A hearing aid comprising:a microphone for generating anelectrical output from sounds external to a user of the hearing aid; anelectrically driven receiver for emitting sound into the ear of the userof the hearing aid, some of the sound emitted by the receiver returningto said microphone to add a feedback to its electrical output, whichfeedback is to be substantially offset; an electronic processor forprocessing the electrical output of the microphone to produce a firstsignal; a first signal combiner for combining the first signal with asecond distinct signal to produce a combined first and second signal forthe receiver of the hearing aid in a proportion wherein the magnitude ofthe second distinct signal is generally less than the magnitude of thefirst signal at loudness of normal speech; and a second signal combinerfor combining the first signal with the second distinct signal toproduce a control signal for the electronic processor wherein themagnitude of the second distinct signal is generally greater than themagnitude of the first signal at loudness of normal speach.
 6. Anelectronic filter for a hearing aid having a microphone for generatingan electrical output from sounds external to a user of the hearing aidand an electrically driven receiver for emitting sound into the ear ofthe user of the hearing aid, some of the sound emitted by the receiverreturning to the microphone to add a feedback to its electrical outputwhich feedback is to be substantially offset, the electronic filtercomprising:a signal generator for generating a probe signal for thereceiver of the hearing aid so that a sound corresponding to the probesignal is included in the sound emitted by the receiver; an electronicprocessor for processing the probe signal in accordance with a series ofcoefficients to produce a first output; a signal combiner for combiningthe first output with the electrical output from the microphone, some ofthe sound corresponding to the probe signal returning to the microphone,to produce a combined signal input having a changing polarity; a signalprocessor for electronically deriving a series of values havingpolarities responsive to the probe signal; a first register for holdingrunning totals and a second register for holding the series ofcoefficients; means for increasing and decreasing each running total insaid first register depending on whether a corresponding value in theseries of values has the same or opposite polarity compared to thecombined signal input; and an adder for respectively adding the runningtotals in said first register to the coefficients in said secondregister to update the coefficients for said electronic processor at arate which is less than a rate at which the increasing and decreasing ofthe running totals in said first register occurs.
 7. An electronicfilter for a hearing aid having a microphone for generating anelectrical output from sounds external to a user of the hearing aid andan electrically driven receiver for emitting sound into the ear of theuser of the hearing aid, some of the sound emitted by the receiverreturning to the microphone to add a feedback contribution to itselectrical output, the electronic filter comprising:a VLSI die; anelectronic processor fabricated on said VLSI die for processing theelectrical output of the microphone to produce a first signal and forcombining the first signal with a second distinct signal to produce acombined first and second signal for the receiver of the hearing aid,said second distinct signal representing noise and being distinct fromthe first signal; and an all-hardware adaptive filter united with andconnected to the electronic processor on said VLSI die for processingthe combined first and second signal while the electronic processor isproducing the first signal, said filter producing an adaptive output tothe electronic processor to substantially offset the feedbackcontribution in the electrical output of the microphone in the hearingaid.
 8. A hearing aid adapted to be coupled to a user, comprising:amicrophone for generating an electrical output from sounds external tosaid user; an electrically driven receiver for emitting sound into theear of said user, some of the sound emitted by said receiver returningto said microphone to add a feedback contribution to its electricaloutput; an electronic processor for processing the electrical output ofthe microphone to produce a first signal and for combining the firstsignal with a second distinct signal to produce a combined first andsecond signal for the receiver means, said second distinct signalrepresenting noise and being distinct from the first signal; and anadaptive filter, having an input coupled to said first signal and thesecond distinct signal, said adaptive filter for processing saidcombined first and second signal while said electronic processor isproducing the first signal to continuously simultaneously offset saidfeedback contribution in the electrical output of said microphone.
 9. Amethod of operating a hearing aid having a microphone for generating anelectrical output from sounds external to a user of the hearing aid andan electrically driven receiver for emitting sound into the ear of theuser of the hearing aid, some of the sound emitted by the receiverreturning to the microphone to add a feedback contribution to itselectrical output, the method comprising the steps of:processing theelectrical output of the microphone to produce a first signal; combiningthe first signal with a second distinct signal to produce a combinedfirst and second signal for the receiver of the hearing aid, said seconddistinct signal representing noise and being distinct from the firstsignal; adaptively filtering the combined first and second signal withcoefficients varied only in response to polarities, to produce anadaptive output in the hearing aid; and continuously simultaneouslyoffsetting with the adaptive output the feedback contribution in theelectrical output of the microphone for the processing step.
 10. Amethod of operating a hearing aid having a microphone for generating anelectrical output from sounds external to a user of the hearing aid andan electrically driven receiver for emitting sound into the ear of theuser of the hearing aid, some of the sound emitted by the receiverreturning to the microphone to add a feedback contribution to itselectrical output, the method comprising the steps of:processing theelectrical output of the microphone to produce a first signal;generating a second distinct signal and varying it in magnitude as afunction of the magnitude of the first signal. combining the firstsignal with a second distinct signal to produce a combined first andsecond signal for the receiver of the hearing aid, said second distinctsignal representing noise and being distinct from the first signal;adaptively filtering the combined first and second signal to produce anadaptive output in the hearing aid; and substantially offsetting withthe adaptive output the feedback contribution in the electrical outputof the microphone for the processing step.
 11. A method of operating ahearing aid having a microphone for generating an electrical output fromsounds external to a user of the hearing aid and an electrically drivenreceiver for emitting sound into the ear of the user of the hearing aid,some of the sound emitted by the receiver returning to the microphone toadd a feedback contribution to its electrical output, the methodcomprising the steps of:processing the electrical output of themicrophone to produce a first signal; combining the first signal with asecond distinct signal to produce a combined first and second signalsfor the receiver of the hearing aid, said second distinct signalrepresenting noise and being distinct from the first signal; adaptivelyfiltering during the processing step the combined first and secondsignal to produce an adaptive output; feeding back the adaptive outputto continuously simultaneously offset the feedback contribution in theelectrical output of the microphone for the processing step.
 12. Anelectronic filter for an electroacoustic system having a microphone forgenerating an electrical output from external sounds and an electricallydriven transducer for emitting sound, some of the sound emitted by thetransducer returning to the microphone to add a feedback contribution toits electrical output, the electronic filter comprising:a VLSI die; anelectronic processor fabricated on said VLSI die for processing theelectrical output of the microphone to produce a first signal and forcombining the first signal with a second distinct signal to produce acombined first and second signal, said second distinct signalrepresenting noise and being distinct from the first signal; and anall-hardware adaptive filter united with and connected to the electronicprocessor on said VLSI die for processing the combined first and secondsignal while the electronic processor is producing the first signal,said adaptive filter producing an adaptive output to the electronicprocessor to substantially offset the feedback contribution in theelectrical output of the microphone.
 13. A hearing aid comprising:amicrophone for producing an input signal; a receiver for producingsound, some of the sound emitted by the receiver returning to themicrophone to add a feedback contribution to the input signal; a firstsignal combiner responsive to the input signal and an adaptive signalfor producing a first combined signal; a noise signal generator forgenerating a noise signal; a second signal combiner responsive to thefirst combined signal and the noise signal for producing an outputsignal for driving the receiver; and an adaptive filter responsive tothe output signal for producing the adaptive signal, wherein theadaptive signal continuously simultaneously offsets the feedbackcontribution in the input signal.
 14. The hearing aid of claim 13wherein the adaptive filter further comprises a logic circuit responsiveto the noise signal and the first combined signal for varying the filterparameters of the adaptive filter.
 15. The hearing aid of claim 13wherein the adaptive filter comprises an all-hardware digital filter.16. The hearing aid of claim 13 further comprising a limiter circuitresponsive to the first combined signal for producing a limited outputsignal, wherein the second signal combiner is responsive to the limitedoutput signal and the noise signal for producing the output signal fordriving the receiver.
 17. The hearing aid of claim 13 further comprisinga filter-limit-filter circuit responsive to the first combined signalfor producing a filtered and limited output signal, wherein the secondsignal combiner is responsive to said filtered and limited output signaland the noise signal for producing the output signal for driving thereceiver.
 18. The hearing aid of claim 17 further comprising a VLSI die,wherein the adaptive filter and the filter-limit-filter circuit arefabricated on the VLSI die.
 19. The hearing aid of claim 13 furthercomprising a VLSI die, wherein the adaptive filter is fabricated on theVLSI die.
 20. The hearing aid of claim 13 further comprising acontroller responsive to the first signal combiner for varying the noisesignal in magnitude as a function of the magnitude of the first combinedsignal.
 21. The hearing aid of claim 13 wherein the adaptive filterfurther comprises a logic circuit responsive to the output signal andthe first combined signal for varying the filter parameters of theadaptive filter.